VoIP-gateway in PSTN <-> SIP mode / outgoing to PSTN

Recently I tried to use the D-Link DVG-6004S as a VoIP-gateway in PSTN <-> SIP mode. Yes, again unsuccessfully but for outgoing (calls) to PSTN.

3CX IP:192.168.10.31
VoIP gateway IP:192.168.10.245
Also VoIP gateway has another Ethernet (NAT-enabled) interface with assigned IP:192.168.1.3. But I don't use that interface at all. D-Link called this interface "WAN".

I made a little research by network monitor and found that 3CX sent SIP-packets to IP:192.168.1.3 during transfer call to PSTN. Then I researched that VoIP gateway registered on the 3CX with SIP-packet consisted "Contact: <sip:10000@192.168.1.3:5060>".

I just was wondering why 3CX sent SIP packets to IP:192.168.1.3. In the 3CX web-interface the

I think, the explanation in the documentation is absolutly unclear:
** part of 3CXPhoneSystemManual30.pdf
This option specifies which IP address/addresses to use in the Contact field when sending SIP messages to this VoIP provider. Default behaviour is to use the STUN resolved IP address. It is possible to specify the internal address instead, or to specify both addresses if the VoIP provider can process this type of information. Additionally, it is possible to specify a custom IP address as the source IP for outgoing SIP messages.
**

Here is part of the Server Status log and the "(192.168.1.3:9000)" substrings are very suspicious for me. This non-usable IP:192.168.1.3 are used again by 3CX.
***
18:00:13.968 StratLink:nHangUp Call(C:A): Ln:10000@d-link hung up call; reason: BYE
18:00:08.125 gt;>:Can't find source interface to use
18:00:04.109 gt;>:Can't find source interface to use
18:00:02.078 gt;>:Can't find source interface to use
18:00:01.046 gt;>:Can't find source interface to use
18:00:00.578 gt;>:Can't find source interface to use
18:00:00.484 CallLegImpl:nConnected Call(C:A): Created audio channel for Ln:10000@d-link (192.168.1.3:9000) with third party (192.168.10.20:49158)
18:00:00.484 StratInOut:nConnected Call(C:A): Setup completed for call from Ext.100 to Ln:10000@d-link
18:00:00.484 CallLegImpl:nConnected Call(C:A): Created audio channel for Ext.100 (192.168.10.20:49158) with third party (192.168.1.3:9000)
17:59:58.578 CallConf:nIncoming Call(C:A): Incoming call from Ext.100 to sip:0019@192.168.10.31
***

3CX Support

Here is part of the Server Status log and the "(192.168.1.3:9000)" substrings are very suspicious for me. This non-usable IP:192.168.1.3 are used again by 3CX.

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This is what your DLink provided to 3CX as destination for RTP stream. We should send to it, we can not send audio to anywhere else.
BTW, are you working for KGB? Any mentioning about IPs makes you suspicious. Mind, that we don't dig thru your PC, all IPs shown are either received from deviced, or specified by you in settings.

Archie
3CX Sr. Coder

3CX News, Tips and How to's at http://www.3cx.com/blog/
Help at http://www.3cx.com/support/

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Please be aware that most gateway devices are designed to interface to VOIP providers, rather than to an IP PBX, so it is commonplace to find that the gateway will send SIP packets out using the WAN port. This effectively makes it almost impossible (for most devices) to use the devices' VOIP-to-IPPBX and NAT services simultaneously.

If the device is connected to the IPPBX network using the LAN port, and the Management Console is set up with the IP address of the gateway's WAN port, this sort of behaviour is typical.

Hope this helps...

Kevin Attard Compagno
Technical Support Manager
http://www.3cx.com

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Achieves connect equipment DVG-6004S with 3CX and registered I can make calls but not receive only have 2 ports Team. The message attached console :

[CM010001] Line configuration does not allow identification of the source of this call. Check Advanced Options for Gateways and Providers and match a field to a correct value from the following: Contact IP: '192.168.1.5'; From: 'FXO'; To: '#100'; Rline: '#100'; Contact: 'FXO'